Asterisk 16.1.0
I'm using hagi and SRV records for a "high availability" configuration
of AGI servers. When the first AGI server in the list is completely
down, asterisk quickly moves on to the next one. That is all good.
My concern is what will happen if asterisk can actually connect to
> On Jan 7, 2019, at 12:25, Joshua C. Colp wrote:
>
> On Mon, Jan 7, 2019, at 1:23 PM, Matt Riddell wrote:
>> Hiya,
>>
>> I would have expected this to show the channels in the bridge inside
>> the anonymous function - it shows the bridge is empty though?
>>
>>var
> On Jan 7, 2019, at 12:25, Joshua C. Colp wrote:
>
> On Mon, Jan 7, 2019, at 1:23 PM, Matt Riddell wrote:
>> Hiya,
>>
>> I would have expected this to show the channels in the bridge inside
>> the anonymous function - it shows the bridge is empty though?
>>
>> var
On Mon, Jan 7, 2019, at 1:23 PM, Matt Riddell wrote:
> Hiya,
>
> I would have expected this to show the channels in the bridge inside
> the anonymous function - it shows the bridge is empty though?
>
> var bridge = ari.Bridge();
> bridge.create({
Hiya,
I would have expected this to show the channels in the bridge inside the
anonymous function - it shows the bridge is empty though?
var bridge = ari.Bridge();
bridge.create({
type: 'holding',
Reply to self: Found the problem after reading this post:
http://lists.digium.com/pipermail/asterisk-dev/2010-March/042735.html
You need to set timert1 in the peer config to *something*, otherwise it
will ignore the timerb setting. Bug? It now looks like this and works fine:
[peer01]
Dear list,
Asterisk 11.25.0 user here. I'm trying to set up failing over to a
second SIP peer if the first SIP peer doesn't answer on our SIP INVITE
within 2 seconds.
In sip.conf I set timerb=2000 for this peer, but it doesn't seem to have
any effect. The timeout is 6.5 seconds instead,
On Mon, Jan 7, 2019, at 3:04 AM, Stefan Viljoen wrote:
> Hi guys
>
> A few months ago I upgraded most of my Asterisk servers to 13 from 1.8.
> I've still got about 25% of my servers on 1.8.
>
> I've since noticed that ringtime on Asterisk 13 - the time difference
> between "start" and "answer"
